Left in target are just .cc files with .h files used externally.
Bug: webrtc:14775
Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41835}
Move code from P2PTransportChannel to Candidate, where we set the
foundation value for remote prflx candidates.
Bug: none
Change-Id: I7dbcb85bca35dca7297136b0706092dd8d2b153c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339902
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41774}
Candidate::type() is currently how the name of the type is fetched,
but that getter returns a non-standard type name.
Instead, I'm adding a new getter, type_name(), will follow up with
updating dependent code that needs the string, to use type_name (and
adapt to potential dependency on "local" or "stun") and then switch
type() to be enum based.
Also adding a test file for Candidate with a couple of basic tests to
start with.
Bug: webrtc:15846
Change-Id: I9b78b2405a9f962a3c07eaa8e72a79854c6f5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41740}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
Using the Api, BWE components are recreated and new settings can be
applied. Initially, the only configuration available is allowing BWE probes without media".
Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration.
Integration test in https://webrtc-review.googlesource.com/c/src/+/337322
Bug: webrtc:14928
Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41687}
This completes the breakup of the rtc_p2p target.
Remaining cleanup is to delete the rtc_p2p target and make clients
depend on the base targets.
Bug: webrtc:15796
Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41683}
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/
Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
This CL does:
- Run IWYU on the relevant elements
- Make connection depend on port_interface, not port
- Make port_allocator depend only on port
- Move some constants from port.h into p2p_constants
This allows a dependency graph without ugly groups.
Bug: webrtc:15796
Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41618}
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.
One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.
Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
This is a reland of commit 79ac694d9b70fa9cd7b6a0f00bbee5d7fbbe64de
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
# Only unrelated bot failures
NOTRY=True
Bug: webrtc:15164
Change-Id: I0f1991409326679a260cb24907808eaa28385350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41603}
This reverts commit 79ac694d9b70fa9cd7b6a0f00bbee5d7fbbe64de.
Reason for revert: Breaks downstream (found another use of
ValueOrDefault instead of value_or)...
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
Bug: webrtc:15164
Change-Id: Ice3f44057b82a7ba9be000d9a0b714152fd07d2f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335701
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41595}
The moment we've all been waiting for.
Step 1: Add type alias (this CL).
Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
Step 3: Delete type alias.
Bug: webrtc:15164
Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41593}
The plan is to replace Members() with Attributes() instead.
For backwards-compatability during the transition, Attribute implements
RTCStatsMemberInterface but the two classes serve the same purpose
which is to allow iterating all metrics of a stats object.
The reason for moving away from "members" is that we already have a way
to express a variable that maybe has a value: absl::optional<T>. The
only information the member adds is the const char* name(), which we'll
move to Attribute in a future CL.
We don't need to maintain an RTCStatsMemberInterface::Type enum in the
future because absl::variant<T> has absl::holds_alternative<T>.
Step 1: Add Attributes().
Step 2: Migrate to Attributes() and delete Members().
Step 3: Replaces all uses of RTCStatsMember<T> with absl::optional<T>
and delete RTCStatsMember + RTCStatsMemberInterface.
Bug: webrtc:15164
Change-Id: I3fdd5b24214bb5cc340a54a0171df73b516e1803
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41507}
Add ulpfec and flexfec to list of resiliency mechanisms taken
into account and in general exclude Comfort Noise (CN) from media
codecs.
Also introduce RtpCodecCapability::IsMediaCodec & ::IsResiliencyCodec
behaving like the MediaCodec methods.
BUG=webrtc:15396
Change-Id: I79041898928190bfdd33a06d8f6975d7556c46b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330424
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41485}
To allow custom FecController use propagated rather than global field trials
note that there is one FecControllerFactory per peer connection factory,
but FecController is created per peer connection and may use per peer connection field trials.
Bug: webrtc:10335
Change-Id: Id25bfaf4b49d4f6d551730c8fd55596ddc49ab47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41478}
while cleaning up Call factory function,
- pick rtp_transport_controller_send_factory based on presence in the config instead of based on the call site thus removing one extra factory function.
- when Call is created through test helper TimeControllerBasedFactory use original media factory instead of direct factory, thus allow to configure degraded call through field trials in tests, and ensure difference with production code path stay minimal in the future.
Bug: webrtc:15656
Change-Id: If9c2a9fc871e139502db2bec0a241d8d64c53720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41329}
Replace CallFactory class with a factory function
Bug: webrtc:15574
Change-Id: Ib1d8cff8d7550da3af01693a7bc117a7bd342258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330000
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41321}
To be submitted after downstream usage has been removed, but no earlier than December 1, 2023.
Bug: webrtc:12598
Change-Id: Id9acbac591c48c0c5883fe8f06cf6a68471b70f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323004
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41290}
This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation.
This poison extends and thus replaces default task queue poison.
Bug: webrtc:15656
Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41247}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
To replace CreateTimeControllerBasedCallFactory
Update webrtc tests to use this new function
Bug: webrtc:15574
Change-Id: I2b74cd930ecc4f72dd1e7aa853764ca298b66ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325527
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41076}
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
instead of requiring to pass in call_factory and media_engine
webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies
Bug: webrtc:15574
Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41049}
This CL just moves the definition and adds a forward.
Actually using the new definition is left for later CLs.
Bug: webrtc:15622
Change-Id: I6d97ef45b98f9eb193c59dd7f8a89c99cfe0ba9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325381
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41048}
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.
The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.
Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262
WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.
This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.
Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
This is part of the long term plan to stop using pointer + length
to pass around buffers.
Bug: webrtc:14870
Change-Id: Ibaf5258fd326b56132b9b5a8a6b1563a763ef2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40512}
Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression.
This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet.
Bug: webrtc:14852
Change-Id: I967160069055036f93e595d328c4d5f1ca483be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39840}
This will clone an encoded audio frame into a sender frame.
Bug: webrtc:14949
Change-Id: Ie62d9f5ec457541b335bde8f2f6e9b6d24704cf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39480}
This reverts commit 8bf321062973939ef35f529640f5e69852e89a7e.
Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9())
Original change's description:
> Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
>
> This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084.
>
> Reason for revert: Breaks upstream project
>
> Original change's description:
> > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
> >
> > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
> >
> > Also default-initialized VideoFrameMetadata::ssrc_ to 0.
> >
> > Bug: webrtc:14708
> > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> > Commit-Queue: Tove Petersson <tovep@google.com>
> > Reviewed-by: Tony Herre <herre@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39411}
>
> Bug: webrtc:14708
> Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39413}
Bug: webrtc:14708
Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39418}
